Transporting voice over asynchronous transfer mode (ATM) or Internet Protocol (IP) based packet networks has several advantages compared to the predominant circuit-switched or Time-Division-Multiplexed (TDM) method. The ability to share the same physical network infrastructure with data traffic; increased network utilization due to statistical multiplexing of voice channels; the potential of creating new services; and the lower cost of voice achieved by using data-traffic-based commodity components are all strong motivations for moving voice to packet networks. However, the use of packet networks for carrying voice channels introduces larger delays, and more importantly, introduces the possibility of packet loss in network routers. Such packet loss may severely impair the quality of the voice conversation.
Designing a carrier-grade, voice solution for a packet network demands that the reliability and quality of voice conversation that has been achieved in the current circuit-switched Public Switched Telephone Network (PSTN) be equaled or exceeded. Practically, achieving this goal requires bounding the probability of any packet loss on a link to an extremely low value, such as a value less than 10−12. Bounding the packet loss rate is achieved by setting the router queues, or buffers, large enough so that no packet loss is expected. Unfortunately, setting the queue length to too large a value increases the cost of the router and results in over-estimation of the queuing delay, which may exceed telecommunications engineering requirements. Accordingly, there is a need for a way to determine a router's queue size to avoid packet loss and minimize the cost of the router.